Efficient data transmission over digital telephone networks using multiple modulus conversion

ABSTRACT

A method of encoding data into a digital sequence to be transmitted over the DTN so as to cause the DTN&#39;s codec to generate a multiple modulus M-ary signal in a manner that will facilitate efficient data transmission and recovery (decoding) by the distant end receiver, even in the presence of robbed-bit signaling (RBS). The preferred embodiments provide an apparatus and method of converting blocks of binary data to a corresponding block of M 1  -ary, M 2  -ary, . . . , M n  symbols using Multiple Modulus Conversion (MMC) to maximize the data rate, while minimizing the required Signal-to-Noise Ratio (SNR) to achieve a desired error rate, in a system having a transmitter connected to the DTN via direct digital access and a receiver connected over a conventional analog subscriber line. A subset of codec codewords is used to represent the M-ary signals. For each time slot (symbol time) one of M separate octets are selected for transmission by the encoder, and the encoder&#39;s output is sent through the DTN to a subscriber loop codec. The value of M can vary among the different time slots. The analog output of the codec corresponds to M-level, or M-ary pulse amplitude modulation, because each of the transmitted octets is converted to one of M analog voltages at the DTN&#39;s codec before being communicated over the subscriber loop.

RELATED APPLICATIONS

This application claims the benefit under 35 U.S.C. §119(e) of U.S.Provisional Patent Application Ser. No. 60/030,843, filed Oct. 15, 1996,entitled "Efficient Data Transmission Over Digital Telephone Networksusing Multiple Modulus Conversion" for all common subject matterdisclosed therein; this application is a CIP of and claims the benefitunder 35 U.S.C. §120 of U.S. patent application Ser. No. 08/888,201,filed Jul. 7, 1997, now U.S. Pat. No. 5,995,548 entitled "ImprovedSignaling Method Using Multiple Modulus Shell Mapping" for all commonsubject matter disclosed therein.

BACKGROUND OF THE INVENTION

This invention relates to data communication over Digital TelephoneNetworks (DTN). In particular, the invention relates to a signalingsystem used in a data distribution system that consists of a datasource, or server, directly connected to the DTN, without anyAnalog-to-Digital (A/D) conversion or Digital-to-Analog (D/A)conversion, and at the receiving end, a client, or subscriber, connectedto a DTN in the normal fashion.

Presently, typical modems used to communicate over the public telephonesystem (one present standard for such communications is detailed in theInternational Telecommunication Union, Telecommunication StandardizationSector ("ITU-T") Recommendation V.34 (1994)), represent binary data byan analog waveform that is modulated in response to the binary data. Thewaveform is in turn analyzed at a receiving modem to recover the binarydata. For modem signals transmitted over the public telephone system,such analog waveforms are treated by central office switches as if thewaveforms were analog voice signals.

FIG. 1 shows a typical communication system presently used for datacommunication. The basic elements of the subscriber loop connection area modem 62 that is connected by an analog line 64 to a local switch 66,which terminates the switched digital telephone network 60. The modem 62is typically located at the subscriber's premises and includes areceiver 68 and transmitter 70. As shown in FIG. 1 the receiver 68 andthe transmitter 70 are coupled to the analog line 64 by a hybrid 72. Thetransmitter 70 converts input digital data 74 into analog signals thatare passed through the hybrid 72 and transmitted over the analog line 24to the local switch 66. Likewise, the receiver 68 converts input analogsignals, which pass from the analog line 24 through the hybrid 32, intodigital data 36.

At the local switch 66 end of the subscriber loop, analog signals fromthe line 64 are directed through a hybrid 78 to an analog-to-digitalconverter 80. The analog-to-digital converter 80 samples the analogsignals converting them into a digital data stream for transmissionthrough the switched digital telephone network 60. For transmission inthe opposite direction, a digital data stream is applied from thedigital telephone network 60 to a digital-to-analog converter 82. Thedigital-to-analog converter 82 converts the data stream into analogsignals that are passed through to the hybrid 78 to the analog line 64,for transmission to the appropriate subscriber.

Typically, the waveforms are digitized into eight bit octets by a codecA/D converter at the central office, and the octets are transmitted indigital format between central offices until they are converted back toan analog signal by a D/A codec at the central office that is connectedto the receiving subscriber loop. The data rate attainable by a modemoperating in such an environment is limited by numerous factorsincluding, in particular, the codec sample rate and the number andspacing of quantization levels of the codec convertors at the centraloffice switches.

The effect on an analog signal associated with sampling the signalamplitude and representing the sample by one of a finite number ofdiscrete (digital) values is generally referred to as quantizationnoise. Most telephone switches utilize voice codecs that performnonlinear A/D and D/A conversions known as μ-law or A-law conversion. Inthese conversion formats, the 8-bit codec codewords, also referred to asoctets, represent analog voltages that are nonlinearly spaced. This typeof conversion performs well for voice signals intended for a humanlistener (especially when transmitted over a noisy line), but have anegative impact on modulated analog waveforms associated with modems.Specifically, codecs that adhere to these standard nonlinear conversionformats implement nonlinearly spaced quantization levels, and have theeffect of increasing quantization noise which is detrimental to modemsignals.

The method and apparatus described herein is for use in a system forconveying digital data across a digital telephone switch to an analogsubscriber, where a data source at one end of the link is connecteddirectly to the digital telephone network (DTN) without undergoing ananalog-to-digital conversion. At the other end of the link, the analogsubscriber is connected to the telephone network in the standardfashion. This type of system may operate at data rates much higher thansystems whose signals undergo an analog-to-digital conversion of atransmitted analog signal and a subsequent digital-to-analog conversionat the receiving end, due at least in part to an associated decrease inquantization noise. In such a system, data is transmitted through thetelephone switching network in a digital format (via, e.g., T1 lines)and is only converted to an analog voltage when it reaches the centraloffice connected to the subscriber's local loop.

FIG. 2 shows a block diagram of such a data distribution system. Thesystem includes a data source 10, or server, having a direct digitalconnection 20 to a digital telephone network (DTN) 30. A client 40 isconnected to the DTN 30 by a subscriber loop 50 that is typically atwo-wire analog line. The DTN routes digital signals from the datasource 10 to the client's local subscriber loop without any intermediaryanalog facilities such that the only analog portion of the link from theserver to the client is the client's local loop. The analog portion thusincludes the channel characteristics of the local loop transmission lineplus the associated analog electronics at both ends of the line. Theonly D/A converter in the transmission path from the server to theclient is the one at the DTN end of the client's subscriber loop. Forthe reverse channel, the only A/D converter in the path from the clientto the server is also at the telephone company's end of the client'ssubscriber loop.

The communications format used by the invention described herein isknown as pulse amplitude modulation. Essentially, the codecs are used togenerate the varying amplitude pulses that are sent over the subscriberloop. Each octet sent to the subscriber's local loop is converted to ananalog voltage by the DTN's codec. Thus each octet may be used togenerate a desired voltage amplitude during that time interval, untilthe next octet is received (presently, octets are transferred at a rateof 8 K Hz).

An impediment to using the full capacity of a data link that has directdigital access to the telephone system and that uses codec codewords toproduce pulse amplitude modulation is that the DTN may periodicallyintroduce errors in the binary data due to the use of what is known inthe telecommunications industry as "robbed bit signaling" (RBS).Additionally, the noise level may prevent a receiver from being able todistinguish between all of the possible codec output levels.

SUMMARY OF THE INVENTION

Described herein is a method of encoding data into a digital sequence tobe transmitted over the DTN so as to cause the DTN's codec to generatean M-ary signal in a manner that will facilitate efficient datatransmission and recovery (decoding) by the distant end receiver, evenin the presence of robbed-bit signaling (RBS).

The preferred embodiments provide an apparatus and method of convertingblocks of binary data to a corresponding block of M₁ -ary, M₂ -ary, . .. , M_(n) symbols using Multiple Modulus Conversion (MMC) to maximizethe data rate, while minimizing the required Signal-to-Noise Ratio (SNR)to achieve a desired error rate, in a system having a transmitterconnected to the DTN via direct digital access and a receiver connectedover a conventional analog subscriber line.

A subset of codec codewords is used to represent the M-ary signals. Foreach time slot (symbol time) one of M separate octets are selected fortransmission by the encoder, and the encoder's output is sent throughthe DTN to a subscriber loop codec. The value of M can vary among thedifferent time slots. The analog output of the codec corresponds toM-level, or M-ary pulse amplitude modulation, because each of thetransmitted octets is converted to one of M analog voltages at the DTN'scodec before being communicated over the subscriber loop. The set of Mlevels is also referred to as a signal constellation, with the Mdifferent voltage levels referred to as signal points within theconstellation.

This approach allows for tighter constellation packing and constellationbalancing, so that a minimum number of constellation points are requiredfor a given bit capacity, and the error rate is not dominated by any onesymbol interval. The net result is a lowering of the required SNR toachieve a desired error rate.

BRIEF DESCRIPTION OF THE DRAWINGS

The preferred embodiment of the present invention is described withreference to the drawings wherein:

FIG. 1 is a block diagram of a standard modem communication system thatutilizes the DTN;

FIG. 2 is a block diagram of a direct digital access communicationsystem;

FIG. 3 shows the conversion between linear samples and μ-law codewordsfor codewords 128-256;

FIG. 4 shows the format of μ-law codewords;

FIG. 5 shows a flow chart of one embodiment of Multiple ModulusConversion; and

FIG. 6 shows a Multiple Modulus Converter.

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT

In the system of FIG. 1, the server 10, having direct digital access tothe DTN may be a single computer, or may include a communications hubthat provides digital access to a number of computers or processingunits. Such a hub/server is disclosed in U.S. Pat. No. 5,528,595, issuedJun. 18, 1996, and is incorporated herein by reference. Anotherhub/server configuration is disclosed in U.S. Pat. No. 5,577,105, issuedNov. 19, 1996, which is also incorporated herein by reference.

In the system shown in FIG. 2, digital data can be input to the DTN as8-bit bytes (octets) at the DTN's clock rate of 8 kHz. At the DTN'sinterface to the subscriber loop, the DTN's codec converts each byte toone of 255 analog voltage levels (two different octets each represent 0volts) that are sent over the client's subscriber loop and received by adecoder at the client's location. As shown in FIG. 3, the analogvoltages, or points, corresponding to the quantization levels arenon-uniformly spaced and follow a generally logarithmic curve. In otherwords, the increment in the analog voltage levels produced from onecodeword to the next is not linear, but depends on the mapping as shownin FIG. 3. Note that the vertical scale of FIG. 3 is calibrated inintegers from 0 to 32,124. These numbers correspond to a linear 16 bitAID converter. The table of FIG. 3 therefore generally represents thelinear to μ-law conversion. As is known to those of ordinary skill inthe art, the sixteenth bit is a sign bit which provides integers from 0to -32124 which correspond to octets from 0 to 127, not shown in FIG. 3.Thus FIG. 3 can be viewed as a conversion between the logarithmic binarydata and the corresponding linear 16-bit binary data. It can also beseen in FIG. 3 that the logarithmic function of the standard conversionformat is approximated by a series of 8 linear segments.

The conversion from octet to analog voltage is well known, and is basedon a system called μ-law coding in North America (and A-law coding inEurope). Theoretically, there are 256 points represented by the 256possible octets, or μ-law codewords. The format of the μ-law codewordsis shown in FIG. 4, where the most significant bit b₇ indicates thesign, the three bits b₆ -b₄ represent the linear segment, and the fourbits, b₀ -b₃ indicate the step along the particular linear segment.These points are symmetric about zero; i.e., there are 128 positive and128 negative levels, including two encodings of zero. Since there are254 points not including zero, the maximum number of bits that can besent per signaling interval (symbol) is just under 8 bits. Otherfactors, such as noise, digital attenuation (pads), channel distortionintroduced by the subscriber loop, and the crowding of points at thesmaller voltage amplitudes and the associated difficulty indistinguishing between them at the decoder/receiver, may reduce themaximum attainable bit rate.

Within the DTN, a supervisory signaling technique, called Robbed BitSignaling ("RBS"), is utilized for call/data routing and control. On anRBS link, a least significant bit (lsb) of certain octets isappropriated, or "robbed", by the DTN in a periodic manner, and used toconvey control information within the telephone system. The octets fromdifferent channels are typically framed in groups of twelve fortransmission between switches within the DTN. On an RBS link, a centraloffice switch robs the lsb of the codewords within every 6th and 12thtime slot within each frame. The robbed bits are used to form asignaling, control and status channel to pass information betweentelephone network equipment. The customer whose data is carried by theDTN loses use of the lsb during those time slots.

Ordinarily, these channels are used for voice communications, and thebit robbing merely increases quantization noise of the effected timeslots. This resultant distortion is barely perceptible to the human earand thus is an acceptable technique of signaling. However, the abovedescribed data transmission system will in general suffer anunacceptable level of data errors on RBS circuits unless some codingmethod is devised to deal with the robbed bit problem. Furthermore, itis possible that more than one interval out of 6 has a bit robbed, ifthere are multiple hops over un-synchronized links. This invention wasalso developed to improve data communication performance on certaintelephone network connections that utilize the RBS supervisory signalingtechnique.

A straightforward method for maintaining acceptable data error rates inthe presence of RBS is to reduce the number of available points byone-half, such that the ambiguity introduced during RBS time slots isresolved at the receiver by interpreting two different received levelsas the same transmitted code. In other words, of the 256 availablepoints or codewords, only 128 of them would be used. At the receiver,two possible analog voltages may be received for each octet transmitteddue to the ambiguity in the lsb. Each of these analog voltages would bedecoded as the same symbol.

Unfortunately, this technique has the disadvantage of sacrificing onebit per symbol/per RBS interval, or 8 kbits/second/RBS interval. A moresophisticated and less costly technique is to encode the data such thatthe number of encoding levels is reduced only during the RBS time slotswithin a frame. This means that the encoder needs to know the positionsof the robbed bit time slots, and needs to be capable of mapping binarydata to fewer points during those slots than during the non-RBS slots.The preferred embodiments of the present invention relate to encodingand mapping the data at the server in a manner that minimizes the datarate penalty due to RBS, while minimizing the required SNR performance.

In order to provide flexible data rates, a preferred embodiment providesa method of sending a fractional number of bits per symbol, so as toachieve maximum data rate and minimum required SNR. In addition, it isdesirable to mitigate the effects of RBS, such that the highest possibledata rate is achieved. Both these requirements can be met with modulusconversion.

Constellations consisting of M_(i) points are selected to meet thefollowing three criteria: ##EQU1## where K is the number of user bits tobe transmitted in an n symbol frame. 2. Points in constellations used inrbs intervals, should be chosen such that the lsbs in theircorresponding μ-law codes are least effected by RBS. (Note theseconstellations will always be smaller than the ones used in non-rbsintervals, since less data is sent during this time.)

3. The probability of symbol error is minimized and balanced. Minimizingthe probability of symbol error is accomplished by choosing points withmaximal spacing, while constraining the points to have some specifiedaverage power. In addition, the number of points with minimum spacingshould be minimized (smallest number of nearest neighbors). Balancingthe symbol error probability, means that mutually, the M, should haveabout the same symbol error rate, since then the overall error rate(frame or block error rate, for example) will not be dominated by thesymbol error rate of any one constellation.

The encoder first collects K bits to send during the n symbol frame. Thefollowing process, called Multiple Modulus Conversion, allows the K bitsto be mapped to n indices, representing K/n bits/symbol (not necessarilyan integer).

The next step of the conversion process is to represent the K bits as aninteger, R, where:

    R=b.sub.0 +b.sub.1 2+b.sub.2 2.sup.2 + . . . +b.sub.K-1 2.sup.K-1.(eq. 2)

In eq.2, b₀ is the lsb and b_(K-1) is the msb (most significant bit) ofthe K-bit data block. The number R may be represented as:

    R=r.sub.1 +r.sub.2 M.sub.1 +r.sub.3 M.sub.1 M.sub.2 + . . . +r.sub.n M.sub.1 M.sub.2 . . . M.sub.n-1.

The result of the modulus conversion process is the production of thevalues r_(i), i=1, . . . n. where the values of the r_(i) areconstrained to the interval: 0≦r_(i) <M_(i). The n values of r_(i)correspond to codec codewords that will be converted to analog levels bythe DTN D/A codec.

A simple algorithm can be used for generating the r_(i), given the M_(i)as follows:

For i=1, . . . , n

R=R/M_(i)

Q=Int(R)

F=R-Q

r_(i) =M_(i) *F

R=Q

Next i

The r_(i) are indices to symbols that are conveyed by the encoder/serverto the receiver by transmitting to the DTN the codewords correspondingto the appropriate points. The indices may be generated in otherfashions, using similar algorithms, without departing from the spiritand scope of the invention.

FIG. 5 depicts the steps performed in converting the data bits tosymbols. At step 100 the K bits are collected for transmission throughmodulus conversion. At step 110 the bits are converted to an integervalue. At step 120 the integer is converted to multiple modulus indicesby using the moduli M_(i) values. At step 130, the multiple modulusindices are transmitted to the telephone network in the form of thecorresponding codec codewords.

At the decoder, the process of decoding or recovering the data involvesidentifying the correct symbol indices r_(i), from the analog voltagessent over the receiver's subscriber loop and then unmapping theseindices through Reverse Multiple Modulus Conversion. One preferredmethod of decoding the symbols is to first restore the integer R fromthe recovered indices r_(i). The following algorithm may be used:

R=0

For i=1, . . . , n

R=M_(n-i+1) *R+r_(n-i+1)

Next i

From R, the K bits in the frame can be recovered.

According to the preferred embodiment of the present invention, thevalues of the M_(i) and their corresponding constellation sets areultimately determined in large part by the SNR of the channel. One ofordinary skill in the art can appreciate that the SNR can be determinedin a number of ways, for example during an initialization period, asequence of known codewords may be transmitted to produce a knownsequence of symbols by the DTN's codec, and the variance from theexpected symbols (points) is measured. Once the SNR is determined, asearch method can be employed to determine the M_(i) 's and theircorresponding constellations. The constellations are chosen from the DACcodes that satisfy the three criteria cited above. That is, search for nsets of points, corresponding to n symbols per frame, and having nmoduli, which simultaneously minimizes the desired probability of symbolerror, while satisfying the data rate criterion 1. One can use thefollowing expression for a good approximation to the probability ofsymbol error when performing the constellation search: ##EQU2## where##EQU3## N.sub.Δ is the number of points with minimum spacing Δ, and σis the noise standard deviation. Also, one needs to account for thenumber of points with spacing Δ, which can be handled as amultiplicative factor times the probability of error.

One advantage of MMC is that it allows a non-integer number of bits tobe mapped to each symbol, which increases efficiency, because theconstellation sizes are not restricted to powers of two (i.e, afractional number of bits/symbol are allowed). As a specific example, ina system having one RBS link, the block of data is converted to amulti-digit modulo-M₁ number (where each digit represents an M₁ -arysymbol) followed by a single digit modulo-M₂ number (representing anadditional M₂ -ary symbol). The concatenated symbols, or digits of themodulo-M₁ and modulo-M₂ numbers, are then transmitted in binary formatas eight-bit bytes, or octets, through the telephone system where eachoctet represents a symbol. At the local loop on the receiver end of thelink, the telephone system converts the octets to analog voltagescorresponding to the M₁ -ary and M₂ -ary symbols. The block of symbolsmay then be decoded by a reverse modulus conversion to recover thebinary data.

FIG. 6 depicts a modulus converter 200 for encoding information bits fortransmission. The converter includes an input buffer 210 for accepting ablock of data bits received on line 220, and at least one modulus buffer230 for storing the modulus values 240. Alternatively, multiple buffersmay be provided in the case of multiple modulus conversion. A processingunit 250 is connected to the input buffer 210 and the modulus buffer(s)230. The processing unit 250 converts the data bits to symbols using themodulus value(s) 240. The converter 200 then outputs the symbols to thetelephone system on output 130. The output 130 is preferably a port of amicroprocessor. The modulus converter preferably has six modulusbuffers, one for each modulus value (corresponding to each time slot ofa six-slot frame). The modulus converter's processing unit 250 ispreferably a microprocessor, or digital signal processor. The buffers210 and 230 are preferably random-access memory (either on-chip, or inassociated RAM chips). Alternatively, the conversion function may beimplemented in an application specific integrated circuit, havingoptimized conversion logic and hardware buffers.

A further embodiment of the modulus conversion method involves combiningmodulus conversion with a technique commonly called shell mapping, andthe combined method may be referred to as Multiple Modulus Shell Mapping(MMSM). MMSM provides a frame mapping technique that uses MultipleModulus Conversion (MMC) and Shell Mapping (SM) to map data bits to asequence of data symbols, or points. The MMSM apparatus includes a shellmapper to generate ring indices from a first block of K data bits, and amodulus converter to select the signal points from within the ring basedon a block of B data bits. MMSM permits the use of constellations havingany integer number of points per ring. MMSM also accommodates variationsin the constellations from time-slot to time-slot within a frame. Thenumber of rings in each constellation preferably remains constant, butthe moduli vary. MMSM produces d_(min) equal to the best of MMC and SM,and in some cases d_(min) may be better than that for either MMC or SM.Further details of MMSM may be found in U.S. patent application Ser. No.08/888,201, entitled "Improved Signaling Method Using Multiple ModulusShell Mapping", filed Jul. 7, 1997, the contents of which areincorporated herein by reference.

A further embodiment of modulus conversion method involves the use of aDC compensation scheme for preventing a DC offset voltage fromaccumulating. Generally, the method includes defining a frame thatincludes at least two unsigned codewords generated by multiple modulusconversion, i.e., the codewords do not include sign bits. An unsignedcodeword is then identified within the frame by applying a rule to thecodewords, for example, the largest magnitude codeword may be selected.Next, a sign bit is appended to the identified unsigned codeword,thereby producing a DC compensating codeword. The sign bit may beselected based upon a weighting function applied to the linear valuesassociated with the previously transmitted codewords. The remainingunsigned codewords have sign bits appended from a pool of user databits.

Additionally, an encoder having an improved DC compensator is provided.The encoder includes a converter that is coupled to the DC compensator.The DC compensator includes a storage device for storing a stream ofunsigned codewords and a sorting device for sorting the storedcodewords. The sorting device is operable to identify a selectedcodeword from the stored codewords in accordance with a rule. The DCcompensator also includes a combiner means for appending a sign bit tothe selected codeword, thereby forming a compensating codeword. Furtheraspects of the DC compensation scheme may be found in U.S. patentapplication Ser. No. 08/871,220, entitled "Frame-Based Spectral ShapingMethod And Apparatus", filed Jun. 9, 1997, the contents of which areincorporated herein by reference.

Preferred embodiments of the present invention have been describedherein. It is to be understood, of course, that changes andmodifications may be made in the embodiment without departing from thetrue scope of the present invention, as defined by the appended claims.

We claim:
 1. A method for transmitting digital information from a datasource having direct access to a digital telephone system for supplyingbinary codewords directly to the digital telephone system, to a receiverconnected to said digital telephone system by an analog subscriber line,wherein the digital telephone system converts the binary codewords fromthe data source to analog voltage levels for transmission to thereceiver, the method comprising the steps of:selecting a number ofsymbol periods, n, in a frame, where n is a multiple of six; selectingat least one block of information bits to be transmitted in said frame,wherein said at least one block comprises K bits; selecting at least oneset of symbols for each of said n symbol periods wherein the number ofsymbols in each said set corresponds to a modulus M of said symbolperiod, and wherein each said symbol corresponds to a binary codeword,and wherein said set of symbols for each said n symbol periods is inpart selected in response to the presence of robbed bit signaling;mapping said at least one block of information bits to said symbols bymultiple modulus conversion; and providing said codewords to the digitaltelephone system.
 2. The method of claim 1 wherein said modulus M variesbetween at least two of said symbol periods.
 3. The method of claim 1wherein n is a multiple of
 6. 4. A method of encoding information bitsfor transmission over a digital telephone system where an encoder hasdirect digital access to the digital telephone system, and wherein saiddigital telephone system appropriates bits during known bit-robbed ratetime slots, comprising the steps of:providing desired moduli values, M₁,M₂, . . . , M_(n) for n time periods wherein said moduli values areselected in response to the presence of robbed bit signaling; providingn sets of binary codewords that define n sets of symbols; mappinginformation bits to said binary codewords selected from said n sets ofsymbols using said moduli values; and providing said codewords to saiddigital telephone system.
 5. The method of claim 4 where the modulivalues of M₁, M₂, . . . , M_(n) are such that ##EQU4## where K is thenumber of bits converted to symbols.
 6. A method of decoding datasymbols transmitted over a digital telephone system where an encoder hasdirect digital access to the digital telephone system for providingbinary octets to the digital telephone system, and wherein said digitaltelephone system appropriates bits from said octets during knownbit-robbed time intervals, and wherein said digital telephone systemconverts the octets to analog voltages, comprising the steps of:defininga transmission frame having transmission time intervals; selecting nsets of symbol codewords where each set corresponds to a particular saidtransmission time interval, and wherein the number of codewords in eachsaid set is selected in response to the presence of robbed bit signalingin the corresponding said transmission time interval; receiving analogvoltages representing symbols to be decoded; and converting saidreceived symbols to a block of binary data using multiple modulusconversion.
 7. A modulus converter for encoding information bits fortransmission over a digital telephone system where an encoder has directdigital access to the digital telephone system, and wherein the digitaltelephone system appropriates bits during known bit-robbed rate timeslots, comprising:an input buffer for accepting a block of data bits; aplurality of modulus buffers for storing a plurality of modulus values,at least one of said modulus values being other than a power of two, andwherein at least one of said modulus values is selected in response tothe presence of robbed bit signaling in the digital telephone system; aprocessing unit connected to said input buffer and said plurality ofmodulus buffers, for converting said block of data bits to symbols usingsaid plurality of modulus values; and an output connected to saidprocessing unit for communication said symbols to the digital telephonesystem.
 8. The modulus converter of claim 7 wherein said plurality ofmodulus buffers comprises six modulus buffers, wherein each of said sixmodulus buffers is adapted for storing a corresponding modulus value. 9.The modulus converter of claim 7 wherein said processing unit is amicroprocessor.
 10. The modulus converter of claim 7 wherein said inputbuffer is a random-access memory register.
 11. The modulus converter ofclaim 7 wherein each of said plurality of modulus buffers is arandom-access memory register.